The present invention relates to a timing phase detector using a four-point fast Fourier transform (hereinafter refer to as FFT), and more particularly, to a timing phase detector used in a timing recovery for converting an analog signal into a digital signal in a digital high-definition television for ground broadcasting using a quadrature amplitude modulation (QAM), or a digital receiver for digital satellite communication or satellite broadcast using a quadrature phase shift keying (QPSK).
A timing phase error estimation algorithm used in a conventional QAM signal mode roughly is divided into a decision-dependence mode and a band-edge component maximization (BECM) mode. The decision-dependence mode samples once for one symbol period so that the sampling speed is fast in a digital communication system in which a transmission speed is rapid. However, the mode requires too many memory devices and much operation, as compared with other modes.
The BECM is to minimize the average square error of the output of an equalizer when a sampled received signal passes through the equalizer. For this, it uses a method of reducing a probability that null is created in the spectrum of a digital signal in which the received signal is sampled with a symbol frequency. The spectral null is produced because the signals of .+-.1/2 a symbol frequency have different phases. Therefore, the BECM is a method of properly controlling the sampling time of the received signal so that the signals of .+-.1/2 the symbol frequency have the same phase The BECM includes a baseband analog mode, band-pass digital mode, and FFT-used mode. At present, the technology development advances toward a trend from analog to digital for the application software integrated circuit (ASIC). The baseband analog mode is out-of-date. The band-pass digital mode involves complicated hardware because it includes a complex-band filter and complex multiplier therein.
The fast Fourier transformation method includes a timing phase error estimation algorithm ("Digital filter and square timing recovery," IEEE Trans. on Comm., Vol. 36, No. 5, pp. 605-612, May 1988) proposed by Oerder and Meyr in 1988, and an algorithm ("A novel frequency domain method for symbol timing recovery and its application in multi-carrier demodulation," Signal Processing IV: Theories and Applications, ed. J. Vandewalle, R. Boite, M. Moonen, and A. Dosterlinck. Amsterdam: Elsevier, 1992) suggested by Barton in 1992. The meyr's replaces a conventional temporal filter with a spectral filter using the Fourier transformation. The block diagram of the algorithm is as shown in FIG. 1.
As depicted in FIG. 1, a baseband analog input signal S1 is applied to an analog-to-digital (A/D) converter 1. The output S2 of A/D converter 1 passes through a square portion 2, FFT portion 3, plane filter portion 4, and cordic (coordinate rotation digital computer) portion 5, and is output as a timing phase error estimation value.
The output S8 of FFT portion 3 contains the phase error component of the signals detached by .+-.1/2 the symbol frequency among input signals. Plane filter 4 makes the output S8 of FFT portion 3 pass directly through a low-band-pass filter in the complex number area, minimizing a hangup that the timing phase detector has. However, the square circuit is hard to implement in hardware. Further, in the circuit the bandwidth of a signal spectrum is doubled eventually to sample four times for every symbol.
In the A-Jalili's, given that the timing phase error is constant, equation (1) can be obtained from two signal components detached as long as the symbol frequency on the spectrum of the received signal. EQU .theta. =1/2.pi.arg(S(f,.theta.)-arg(S(f +1/T.theta.)-arg(H(f+1/T)),(1)
Where T is the period of a symbol;
S(f) is the spectrum of a received signal; and
H(f) is the spectrum of the whole waveform.
In this case, suppose that H(f) is a function with a constant phase, that is, -arg(H(f+1/T)) is 0 or a constant value, the A1-Jalili's algorithm can be exampled as shown in FIG. 2. The received signal S1 is converted into a digital signal S2 in A/D converter 1, and applied to FFT portion 10. The FFT outputs S.sub.5-1 -S.sub.5-4 in accordance with the number of windows of FFT portion 10 are applied to corresponding cordics 11-4 through 11-4 of cordic portion 11 to calculate the phase value. Respective outputs S.sub.6-1 -S.sub.6-4 of cordic portion 11 pass through a subtracting portion 12 of a pair of subtractors 12-1 and 12-2, and an average portion 13 so as to calculate timing phase error estimation value S7.
This method needs no square circuit unlike the Meyr's, and avoids widening the spectral bandwidth of signal so that the overlap of signal spectrum does not occur though sampling is performed twice for every symbol. However, as the window of FFT portion 10 increases, the amount of hardware also swells. Further, several pairs of signal components detached as long as symbol frequency 1/T are used to increase the hardware of cordic portion 11.
For another FFT-related method, U.S. Pat. No. 5,199,078 is known. In this disclosure, a digital audio signal is divided into overlap time window segments for the purpose of FFT. The FFT-transformed segments are grouped in frequencies to obtain the average. This method is advantageous in replacing several-magnitude values with one value. However, the hardware for dividing and processing a signal into several segments is complicated.